_debug* -> _debug

Don't use a 8k buffer to format message, let v* functions do the work
Use yellow for warnings
Use either syslog either console output

daemon/src/audio/alsa/alsalayer.cpp

@@ -111,7 +111,7 @@ AlsaLayer::~AlsaLayer (void)
 void
 AlsaLayer::closeLayer()
 {
-    _debugAlsa ("Audio: Close ALSA streams");
+    _debug ("Audio: Close ALSA streams");
 
     try {
         /* Stop the audio thread first */
@@ -121,7 +121,7 @@ AlsaLayer::closeLayer()
             audioThread_ = NULL;
         }
     } catch (...) {
-        _debugException ("Audio: Exception: when stopping audiortp");
+        _debug ("Audio: Exception: when stopping audiortp");
         throw;
     }
 
@@ -155,10 +155,10 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
 
     audioPlugin_ = plugin;
 
-    _debugAlsa (" Setting AlsaLayer: device     in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_);
-    _debugAlsa ("                   : alsa plugin=%s", audioPlugin_.c_str());
-    _debugAlsa ("                   : nb channel in=%2d, out=%2d", inChannel_, outChannel_);
-    _debugAlsa ("                   : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString);
+    _debug (" Setting AlsaLayer: device     in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_);
+    _debug ("                   : alsa plugin=%s", audioPlugin_.c_str());
+    _debug ("                   : nb channel in=%2d, out=%2d", inChannel_, outChannel_);
+    _debug ("                   : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString);
 
     audioThread_ = NULL;
 
@@ -218,7 +218,7 @@ AlsaLayer::startStream (void)
             audioThread_ = new AlsaThread (this);
             audioThread_->start();
         } catch (...) {
-            _debugException ("Fail to start audio thread");
+            _debug ("Fail to start audio thread");
         }
     }
 
@@ -240,7 +240,7 @@ AlsaLayer::stopStream (void)
             audioThread_ = NULL;
         }
     } catch (...) {
-        _debugException ("Audio: Exception: when stopping audiortp");
+        _debug ("Audio: Exception: when stopping audiortp");
         throw;
     }
 
@@ -404,12 +404,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
 
     int err;
     if ((err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot initialize hardware parameter structure (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot initialize hardware parameter structure (%s)", snd_strerror (err));
         return false;
     }
 
     if ((err = snd_pcm_hw_params_set_access (pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set access type (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set access type (%s)", snd_strerror (err));
         return false;
     }
 
@@ -417,7 +417,7 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
     format = SND_PCM_FORMAT_S16_LE;
 
     if ((err = snd_pcm_hw_params_set_format (pcm_handle, hwparams, (snd_pcm_format_t) format)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set sample format (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set sample format (%s)", snd_strerror (err));
         return false;
     }
 
@@ -427,19 +427,19 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
     unsigned int exact_ivalue = audioSampleRate_;
 
     if ((err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
-        _debugAlsa ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err));
         return false;
     } else
         _debug ("Audio: Set audio rate to %d", audioSampleRate_);
 
     if (dir != 0) {
-        _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , audioSampleRate_, exact_ivalue);
+        _debug ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , audioSampleRate_, exact_ivalue);
         audioSampleRate_ = exact_ivalue;
     }
 
     /* Set the number of channels */
     if ((err = snd_pcm_hw_params_set_channels (pcm_handle, hwparams, 1)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err));
         return false;
     }
 
@@ -449,12 +449,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
     dir = 0;
 
     if ((err = snd_pcm_hw_params_set_period_size_near (pcm_handle, hwparams, &exact_lvalue, &dir)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set period time (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set period time (%s)", snd_strerror (err));
         return false;
     }
 
     if (dir != 0)
-        _debugAlsa ("Audio: Warning: (%i) The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", type, periodsize, exact_lvalue);
+        _debug ("Audio: Warning: (%i) The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", type, periodsize, exact_lvalue);
 
     periodSize_ = exact_lvalue;
     /* Set the number of fragments */
@@ -462,19 +462,19 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
     dir = 0;
 
     if ((err = snd_pcm_hw_params_set_periods_near (pcm_handle, hwparams, &exact_ivalue, &dir)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err));
         return false;
     }
 
     if (dir != 0)
-        _debugAlsa ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue);
+        _debug ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue);
 
     periods = exact_ivalue;
 
     /* Set the hw parameters */
 
     if ((err = snd_pcm_hw_params (pcm_handle, hwparams)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err));
         return false;
     }
 
@@ -487,12 +487,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
     /* Set the start threshold */
 
     if ((err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, periodSize_ * 2)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err));
         return false;
     }
 
     if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) {
-        _debugAlsa ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err));
+        _debug ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err));
         return false;
     }
 
@@ -583,14 +583,14 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
                 handle_xrun_playback (handle);
 
                 if (snd_pcm_writei (handle, buffer , frames) < 0)
-                    _debugAlsa ("Audio: XRUN handling failed");
+                    _debug ("Audio: XRUN handling failed");
 
                 trigger_request_ = true;
 
                 break;
 
             default:
-                _debugAlsa ("Audio: Write error unknown - dropping frames: %s", snd_strerror (err));
+                _debug ("Audio: Write error unknown - dropping frames: %s", snd_strerror (err));
                 stopPlaybackStream ();
                 break;
         }
@@ -616,12 +616,12 @@ AlsaLayer::read (void* buffer, int toCopy)
             case -EPIPE:
             case -ESTRPIPE:
             case -EIO:
-                _debugAlsa ("Audio: XRUN capture ignored (%s)", snd_strerror (err));
+                _debug ("Audio: XRUN capture ignored (%s)", snd_strerror (err));
                 handle_xrun_capture();
                 break;
 
             case EPERM:
-                _debugAlsa ("Audio: Capture EPERM (%s)", snd_strerror (err));
+                _debug ("Audio: Capture EPERM (%s)", snd_strerror (err));
                 prepareCaptureStream ();
                 startCaptureStream ();
                 break;
@@ -639,7 +639,7 @@ AlsaLayer::read (void* buffer, int toCopy)
 void
 AlsaLayer::handle_xrun_capture (void)
 {
-    _debugAlsa ("Audio: Handle xrun capture");
+    _debug ("Audio: Handle xrun capture");
 
     snd_pcm_status_t* status;
     snd_pcm_status_alloca (&status);
@@ -651,20 +651,20 @@ AlsaLayer::handle_xrun_capture (void)
             startCaptureStream ();
         }
     } else
-        _debugAlsa ("Audio: Get status failed");
+        _debug ("Audio: Get status failed");
 }
 
 void
 AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
 {
-    _debugAlsa ("Audio: Handle xrun playback");
+    _debug ("Audio: Handle xrun playback");
 
     snd_pcm_status_t* status;
     snd_pcm_status_alloca (&status);
 
     int state;
     if ((state = snd_pcm_status (handle, status)) < 0)
-        _debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
+        _debug ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
     else {
         int state = snd_pcm_status_get_state (status);
 
@@ -733,9 +733,9 @@ AlsaLayer::getSoundCardsInfo (int stream)
                 snd_pcm_info_set_device (pcminfo , 0);
                 snd_pcm_info_set_stream (pcminfo, (stream == SFL_PCM_CAPTURE) ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK);
 
-                if (snd_ctl_pcm_info (handle ,pcminfo) < 0) _debugAlsa (" Cannot get info");
+                if (snd_ctl_pcm_info (handle ,pcminfo) < 0) _debug (" Cannot get info");
                 else {
-                    _debugAlsa ("card %i : %s [%s]",
+                    _debug ("card %i : %s [%s]",
                                 numCard,
                                 snd_ctl_card_info_get_id (info),
                                 snd_ctl_card_info_get_name (info));

daemon/src/audio/audiortp/AudioRtpFactory.cpp

@@ -58,7 +58,7 @@ AudioRtpFactory::~AudioRtpFactory()
 void AudioRtpFactory::initAudioRtpConfig ()
 {
     if (_rtpSession != NULL) {
-        _debugException ("An audio rtp thread was already created but not" \
+        _debug ("An audio rtp thread was already created but not" \
                          "destroyed. Forcing it before continuing.");
         stop();
     }
@@ -161,7 +161,7 @@ void AudioRtpFactory::stop (void)
     _info ("AudioRtpFactory: Stopping audio rtp session");
 
     if (_rtpSession == NULL) {
-        _debugException ("AudioRtpFactory: Rtp session already deleted");
+        _debug ("AudioRtpFactory: Rtp session already deleted");
         return;
     }
 
@@ -174,7 +174,7 @@ void AudioRtpFactory::stop (void)
         delete _rtpSession;
         _rtpSession = NULL;
     } catch (...) {
-        _debugException ("AudioRtpFactory: Error: Exception caught when stopping the audio rtp session");
+        _debug ("AudioRtpFactory: Error: Exception caught when stopping the audio rtp session");
         throw AudioRtpFactoryException ("AudioRtpFactory: Error: caught exception in AudioRtpFactory::stop");
     }
 }

daemon/src/audio/audiortp/AudioZrtpSession.cpp

@@ -73,7 +73,7 @@ AudioZrtpSession::~AudioZrtpSession()
     try {
         terminate();
     } catch (...) {
-        _debugException ("AudioZrtpSession: Thread destructor didn't terminate correctly");
+        _debug ("AudioZrtpSession: Thread destructor didn't terminate correctly");
         throw;
     }
 

daemon/src/logger.cpp

@@ -45,52 +45,40 @@ void log (const int level, const char* format, ...)
         return;
 
     va_list ap;
-    const char *prefix = "<> ";
-    const char *color_prefix = "";
 
-    switch (level) {
-        case LOG_ERR: {
-            prefix = "<error> ";
-            color_prefix = RED;
-            break;
+    if (consoleLog) {
+        const char *color_prefix = "";
+        switch (level) {
+            case LOG_ERR:
+                color_prefix = RED;
+                break;
+            case LOG_WARNING:
+                color_prefix = YELLOW;
+                break;
         }
-        case LOG_WARNING: {
-            prefix = "<warning> ";
-            color_prefix = LIGHT_RED;
-            break;
-        }
-        case LOG_INFO: {
-            prefix = "<info> ";
-            color_prefix = "";
-            break;
-        }
-        case LOG_DEBUG: {
-            prefix = "<debug> ";
-            color_prefix = "";
-            break;
-        }
-    }
 
-    char buffer[8192];
-    va_start (ap, format);
-    vsnprintf (buffer, sizeof buffer, format, ap);
-    va_end (ap);
+    	fputs(color_prefix, stderr);
 
-    if (consoleLog)
-        fprintf(stderr, "%s%s"END_COLOR"\n", color_prefix, buffer);
+        va_start (ap, format);
+        vfprintf(stderr, format, ap);
+        va_end (ap);
 
-    syslog (level, "%s%s", prefix, buffer);
+        fputs(END_COLOR"\n", stderr);
+    } else {
+		va_start (ap, format);
+		vsyslog (level, format, ap);
+		va_end (ap);
+    }
 }
 
 void setConsoleLog (bool c)
 {
-    Logger::consoleLog = c;
+    consoleLog = c;
 }
 
 void setDebugMode (bool d)
 {
-    Logger::debugMode = d;
+    debugMode = d;
 }
 
 }
-

daemon/src/logger.h

@@ -46,10 +46,6 @@ void setDebugMode (bool);
 #define _info(...)	Logger::log(LOG_INFO, __VA_ARGS__)
 #define _debug(...)	Logger::log(LOG_DEBUG, __VA_ARGS__)
 
-#define _debugException(...)	Logger::log(LOG_DEBUG, __VA_ARGS__)
-#define _debugInit(...)			Logger::log(LOG_DEBUG, __VA_ARGS__)
-#define _debugAlsa(...)			Logger::log(LOG_DEBUG, __VA_ARGS__)
-
 #define BLACK "\033[22;30m"
 #define RED "\033[22;31m"
 #define GREEN "\033[22;32m"

daemon/src/managerimpl.cpp

@@ -123,11 +123,11 @@ void ManagerImpl::init (std::string config_file)
     if (_audiodriver) {
         unsigned int sampleRate = _audiodriver->getSampleRate();
 
-        _debugInit ("Manager: Load telephone tone");
+        _debug ("Manager: Load telephone tone");
         std::string country(preferences.getZoneToneChoice());
         _telephoneTone = new TelephoneTone (country, sampleRate);
 
-        _debugInit ("Manager: Loading DTMF key (%d)", sampleRate);
+        _debug ("Manager: Loading DTMF key (%d)", sampleRate);
 
         sampleRate = 8000;
 
@@ -2708,7 +2708,7 @@ void ManagerImpl::setEchoCancelDelay(int delay)
  */
 bool ManagerImpl::initAudioDriver (void)
 {
-    _debugInit ("Manager: AudioLayer Creation");
+    _debug ("Manager: AudioLayer Creation");
 
     audioLayerMutexLock();
 
@@ -2945,12 +2945,12 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
     unsigned int sampleRate = _audiodriver->getSampleRate();
 
     delete _telephoneTone;
-    _debugInit ("Manager: Load telephone tone");
+    _debug ("Manager: Load telephone tone");
     std::string country = preferences.getZoneToneChoice();
     _telephoneTone = new TelephoneTone (country, sampleRate);
 
     delete _dtmfKey;
-    _debugInit ("Manager: Loading DTMF key with sample rate %d", sampleRate);
+    _debug ("Manager: Loading DTMF key with sample rate %d", sampleRate);
     _dtmfKey = new DTMF (sampleRate);
 
     // Restart audio layer if it was active
@@ -2966,7 +2966,7 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
  */
 void ManagerImpl::initVolume ()
 {
-    _debugInit ("Initiate Volume");
+    _debug ("Initiate Volume");
     setSpkrVolume (audioPreference.getVolumespkr());
     setMicVolume (audioPreference.getVolumemic());
 }